1. Field of the Invention
The present invention relates to a Voice over Internet Protocol (VoIP) terminal having a Quality of Service (QoS) monitoring function and a QoS monitoring method.
2. Description of the Related Art
Generally, VoIP is a protocol for transferring video, voice and FAX messages over the Internet. With VoIP, a user accesses the Internet using a personal computer (PC) or an independent IP-based device to transmit/receive voice, as the user makes a call to a gateway using a conventional public switched telephone network (PSTN) terminal, or to transmit/receive real time media such as video.
A VoIP end-point device (e.g., a gateway, an IP phone, or a PC) of a sender continuously exchanges real-time transport protocol (RTP) voice packets with a recipient to conduct voice communication through RTP on an IP network. The VoIP end-point device transmits and receives QoS information, such as packet loss value, jitter and delay, and RTP control protocol (RTCP) information including call session information, to and from the recipient at uniform intervals.
RTP is a protocol for transferring real-time data, such as audio or video data, over a multicast or unicast network. The RTP does not provide a connection, generally operates as an upper layer of a user datagram protocol (UDP), and uses multiplexing and checksum features of the UDP.
The RTCP periodically transmits a control packet to a recipient using the following messages: a sender report (SR) including transmission/reception statistics information of an active sender; and a receiver report (RR) including transmission/reception information of participants, not an active sender, a source description (SDES) describing a canonical name (CNAME) and a source name, BYE indicating termination of an RTP session, and application (APP) specifying a function confined to the application upon testing a new application or a new function. The RTCP packet has a fixed header field and a variable field having a length varying with the type of packet. For effective information, several RTCP packets may form one UDP packet.
Recently, a Voice over Wireless LAN (VoWLAN; wireless Internet telephone) technology transmitting voice over a WLAN, which is the most widespread wireless technique, has emerged as a new mobile telephone technology, just as VoIP systems have emerged as a new wired telephone technology. This is because the VoWLAN is inexpensive and convenient because of its mobility, in addition to its having the benefits of wired Internet telephone VoIP.
The VoWLAN transmits voice data over the WLAN. Typical Internet telephones operate on a wired network, while the VoWLAN uses the WLAN as a medium.
The VoWLAN can provide convenient voice communication because of its guaranteed mobility within a coverage area of an access point (AP). In addition, the VoWLAN can provide significant cost savings compared to line-based telephoning since it uses a pre-built network. Particularly, the VoWLAN is advantageous for its ability to adapt to the needs of the future, for example, video telephoning.
Since all voice data in the VoIP are composed as RTP packets and are continuously transmitted over a data network, the VoIP needs a certain network bandwidth to enable smooth communication.
Since VoIP technology uses communication channels of a data communication network (Internet), voice data may be affected by delay, jitter, or loss on the network, as normal data is.
For this reason, measurement of QoS in a VoIP service requires diagnosis of a state of the network. In a QoS measuring method in a voice communication system using conventional VoIP technology, a traffic measuring device is connected to a VoIP packet path to directly parse the VoIP packet, or a QoS state monitoring system is developed to collect data.
However, the QoS measuring method in the voice communication system using conventional VoIP technology requires a separate device or system.
Furthermore, since provided values themselves are sub-classified into Round-Trip delay Time (RTT), jitter, and loss, quality of actual VoIP service may depend on a manager's knowledge and experience. In this regard, the delay (RTT) refers to time measured from the moment a receiving terminal receives packet data from a transmitting terminal until the moment the receiving terminal transmits packet data back to the transmitting terminal.
Also, even in the same network state, terminals having different types or characteristics may be influenced differently. As this has not been considered, the conventional art is not capable of providing optimal functions according to the type or properties of a terminal.